Le 11/11/2019 à 19:53, Jakob Laue a écrit :
Okay,
now I found some time to look at Giulios patch and I think I understood it :P 
(So many $'s in there :P)
So, what actually happens in that patch is that on [loadbang] we read 32768 
samples into an array using soundfiler.
Then when the file should be played, we do play the first 32768 samples using 
[tabplay~] and simultaneously load the rest of the file using [readsf~]. Then, 
when [tabplay~] is done playing the first bit of the sample, we send a [1( to 
[readsf~] to start playing the rest of the file.
I don't have any audio pops playing eight samples simultaneously, even using 
the cheap hdmi-to-cinch-converter as my audio interface (which caused more pops 
in comparison to the Scarlett USB interface while trying olivers patch). Also, 
I don't have to adjust Pd's audio settings. I can use the standard settings 
(44100, 64 blocksize, 25 msec) and I don't get audio pops, which is good.

But there is one thing that I realised: The wav-files are not played at 
original speed. They are all recorded in 122 bpm. When I play them with Giulois 
patch, the samples are played quite slowly, at around 111 bpm.

look like the sample are recorded at 48KHz and play at 44100.

In a later version of my sample player, I want to be able to change the 
bpm/tempo in which the samples are played, so I will need to have some 
mechanism for changing this. Using e.g. [phasor~] to play samples, this should 
work, but I don't know if it will be possible to do this using Giulios 
approach. In the help file of [tabplay~], it says that samples are played with 
no transposition, so maybe a transposition will not be possible to implement?! 
On the other hand, it says that [tabplay~] does not have the 
interpolation-artefacts that [tabread4~] has, which I would have, if I used a 
[phasor~] approach.

if you want to change the tempo, you need to somehow interpolate the samples...

best

All the best, Jakob
*Gesendet:* Dienstag, 05. November 2019 um 15:40 Uhr
*Von:* "oliver" <oli...@klingt.org>
*An:* Kein Empfänger
*Cc:* Pd-List <pd-list@lists.iem.at>
*Betreff:* Re: [PD] Raspberry Pi: Loading Samples RAM problem
Jakob Laue wrote:
 > Okay,
 > finally I found some time to try some stuff out. First, I gave Olivers
 > examples a try.
 > I made a patch with eight [ol_sfplay~] objects to simulate my sample
 > player and tested this patch on my old Raspi 2
 > and also a Raspi 3. I also tested two different audio interfaces: a
 > Scarlett 2i4 USB audio interface and a HDMI-to-CINCH-Converter running
 > on the 2835 ALSA driver.
 > The best result was achieved by the Raspberry Pi 3 with the Scarlett
 > audio interface. I used
 > - block length of 64 from Pd's preferences
 > - buffersize of 512 and 1024 for [ol_sfplay~]
 > I have no pops and Pd does not freeze.
 > When I use the same configurations with the hdmi-to-cinch-converter and
 > alsa, I get audio pops.

please also consider using even 2048 as buffersize for [ol_sfplay~]. AND
you might want to increase PD's audio delay with the "-audiobuf" startup
flag.

try "-audiobuf 120" or higher and work your way down from there.

i remember using this method on a PI2 with an 8 track wav-file and
really had to increase those values to get a smooth playback.

it worked alright in the end but then i had no video-traffic ...

in general, use a PI3 if possible

best

oliver



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